similar to: Optional URL param

Is it just my box, or is there something flaky in the implementation of "show manager commands"? Note: I'm using putty. About half way through this, I toggled my KVM over to the desktop and logged in to try and recreate it. The output was the same as the last two entries in this dump. bebop*CLI> show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote:

Hi there, I''ve used Javascript in the past to disable form fields, but I''m wondering if there''s a way to easily do this with Rails during the page generation. In my app, i have a page that has a form. I want to disable the form if a set of URL parameters are not present. I could perhaps do some javascript onload function that disables all the form fields, or add

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

Later versions of gcc include an option in the form of --param=name=value. This patch adds support for this and continues to support the --param name=value option. Signed-off-by: Mike Pagano <mpagano at gentoo.org> --- klcc/klcc.in | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/klcc/klcc.in b/klcc/klcc.in index 711a832..5d00bfd 100644 ---

I'm stuck on queues! The way I read what documentation I have found, if I set up a queue like this: [general] persistentmembers = yes [testq] musiconhold=default strategy = ringall timeout = 10 retry = 5 context = testing member => SIP/100 and then add into extensions something like this: [incomingiax] exten => 1234,1,Dial(SIP/100,10) exten => 1234,2,Queue(testq|tTH|||300)

I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current) Today when I setup queues for the first time (with one member in my default queue), I got some really strange behaviour, aside from my hysterical laughing after hearing the default MOH =) I only have one SIP hardphone I'm testing with right now, so I tested using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog

Hi, I''m trying to create a select box with an onchange event. When onchange is fired, I want to redirect to a nested rest url. The problem is that I don''t know how to pass the selected value to the url: <select id="category" onchange=" <%= remote_function(:url => category_articles_path(selected value?) ) %> "> Any suggestions? -- Posted

Hello I have problem with transfer call if using ACD When i using ACD with agent and queue setting, i cannot monitor call and transfer call. this's my setting - i have 2 IAX phone (phone number as 201, 202), agent.conf agent => 1001,4321,member 1 agent => 1002,4321,member 2 agent => 1003,4321,Tin then, queues.conf [MyQueue] music=default strategy=ringall timeout=30 retry=5

Hi, I've already post this to the forum three days ago, sorry if it's sounds like a crosspost, but I've got no replies, so I'm trying other channels :) This is the link to the forum post if someone prefer to reply here: http://forums.asterisk.org/viewtopic.php?f=1&t=86985 I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging activated. I'm using CEL because

I am trying to get two * boxes to communicate with eachother. I have read http://www.voip-info.org/wiki-Asterisk+-+dual+servers as well as information on IAX channels, the Dial() command, and the switch statement in extensions.conf. But I am having no luck. I have a working * box running with a Zap card that I want to be the server machine. I have another little box running * with just a

Hi, While trying to reproduce: https://bugzilla.redhat.com/show_bug.cgi?id=1325667 I also gave glxgears a quick test with mesa master, this resulted in the following errors being printed to the terminal from which glxgears was started : unknown fragment shader param 17 unknown fragment shader param 17 unknown fragment shader param 17 unknown fragment shader param 17 unknown fragment shader

Hi, When I builded realtek image, the compiler stop in Samba compiling process, Did I miss something? Thanks. Best regards, Wilson ps. error message was list below, make[3]: Entering directory `/media/FD2/Project/WAP-3522/rtl819x-SDK-v3.4.6.7-full-package/rtl819x/users/samba' make -C samba-3.2.15/source make[4]: Entering directory

https://bugzilla.netfilter.org/show_bug.cgi?id=911 Summary: The ICMPv6 type is param-problem or parameter-problem ? Product: nftables Version: unspecified Platform: x86_64 OS/Version: Debian GNU/Linux Status: NEW Severity: normal Priority: P5 Component: nft AssignedTo: pablo at

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Since we upload all parameters on every program / constbuf change, we don't have to reserve space and can just use the whole buffer. Doesn't apply to the buffer holding immediates. --- src/gallium/drivers/nv50/nv50_program.c | 39 ++++++++---------------------- src/gallium/drivers/nv50/nv50_program.h | 6 ++-- 2 files changed, 14 insertions(+), 31 deletions(-) diff --git

Sunday news is today published on a monday. Yesterday was fourth of july, and I used that as an excuse for being off line yesterday. (Sweden's national day is June 6th - and it's not yet a public holiday, btw). Most of my Asterisk time lately have been used for producing the registration site for Astricon and tracking down speakers that haven't sent in their material for the conference